J.Chesterfield
                                                             E.Schooler
                                                   AT&T Labs - Research
   Internet Draft                                                 J.Ott
   Document: draft-ietf-avt-rtcpssm-00   Tellique Kommunikationstechnik
                                                                   GmbH
   Expires: August 2002                                   February 2002

           RTCP Extensions for Single-Source Multicast Sessions
                           with Unicast Feedback


Status of this Memo
   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   Drafts.

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   http://www.ietf.org/ietf/1id-abstracts.txt

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Abstract
   This document specifies a modification to the Real-time Transport
   Control Protocol (RTCP) to use unicast feedback. The proposed
   extension is useful for single source multicast sessions such as
   Source Specific Multicast (SSM) communication where the traditional
   model of many-to-many group communication is either not possible or
   not preferred. In addition, it can be applied to any group that
   might benefit from a sender controlled summarised reporting
   mechanism.


1. Conventions and Acronyms
   The keywords MUST, MUST NOT, REQUIRED, SHALL, SHALL NOT, SHOULD,
   SHOULD NOT, RECOMMENDED, MAY, and OPTIONAL, when they appear in this
   document, are to be interpreted as described in RFC 2119.


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2. Introduction
   The Real-time Transport Protocol (RTP) [1] provides a real-time
   transport mechanism suitable for unicast or Internet Standard
   multicast communication between multimedia applications. Typical
   uses are for real-time or near real-time group communication via
   audio and video data streams. An important component of the RTP
   protocol is the control channel, defined as the Real-Time Control
   Protocol (RTCP). RTCP involves the periodic transmission of control
   packets between group members in a session, enabling the
   distribution or calculation of session specific information such as
   packet loss and round trip time to other hosts, and group size
   estimation. An additional advantage of providing a control channel
   for a session is that a third-party session monitor can listen to
   the traffic to establish network conditions and to diagnose faults
   based on receiver locations.

   RTP was designed to operate in a unicast mode or in the traditional
   multicast mode of Any Source Multicast (ASM) group communication,
   where both one-to-many and many-to-many communication are supported
   via a common group address in the range 224.0.0.0 through
   239.255.255.255. Typical routing protocols that enable such
   communication are the Distance Vector Multicast Routing Protocol
   (DVMRP) [2] or Protocol Independent Multicast (PIM) [3][4] in
   combination with an Inter-domain routing protocol such as Multicast
   Border Gateway Protocol (MBGP) [5] with Multicast Source Discovery
   (MSDP) [6]. Such routing protocols enable a host to join a single
   multicast group address and to send to or to receive data from all
   members in the group with no prior knowledge of the membership. In
   order to enable such a service in the network, however there is a
   great deal of complexity involved at the routing level.

   An alternative approach has been developed for multicast groups with
   just a single sender. The Source Specific Multicast (SSM) [7] model
   has the advantage of removing a great deal of the routing complexity
   involved in multicast group creation and source information
   distribution. The disadvantage of SSM, with respect to real-time
   traffic using RTP, is that the simplification to the routing
   protocols removes the ability for any member of the group to
   communicate with any other member of the group without an explicit
   join to that host.

   The solution proposed in this draft defines a new method for
   distributing control information amongst all members in a multicast
   session and is designed to operate under any multicast group
   communication scenario. It is, however, of particular benefit to SSM
   sessions in the absence of receivers being able to communicate with
   each other directly. The RTP data stream protocol itself is
   unaffected. The basic architectural models to which this feedback
   method could apply include:

   a) SSM groups with a single sender.

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      This is the main motivation behind the unicast RTCP feedback
      mechanism. The proposed extensions allow SSM groups that do not
      have many-to-many communication capability to still receive RTP
      data streams and to continue to participate in the RTP control
      protocol, RTCP. Because SSM adopts the notion of a sender data
      channel that provides a one-to-many communication facility from
      the source to all the receivers in the group, the RTCP feedback
      is unicast to the source on the standard RTCP port.

   b) One-to-many broadcast networks.
      An example of such a network is a satellite network with a
      terrestrial low-bandwidth return channel or a broadband cable
      link. This architecture differs very little from the  SSM channel
      concept, but is likely to require a translator of some kind to
      render the RTP data stream onto the satellite or cable
      distribution channel.

   c) ASM with a single sender.
      An SDP session announcement type may identify a session as having
      a single sender receiving unicast RTCP feedback. Receivers join
      the multicast group address and receive RTP and RTCP data from
      the source on the specified address/port combinations. The RTCP
      feedback is unicast back to the source on the RTCP port. This
      model is not more efficient than a standard multicast group RTP
      communication scenario, and is therefore not recommended to
      replace the traditional mechanism. However it may be help to
      prevent overtaxing multicast routing infrastructure that does not
      scale as efficiently.

   SSM sessions are typically assigned a value in the group address
   range 232.0.0.0 through 232.255.255.255, although this is not a
   requirement. A session may be assigned any valid multicast address,
   as long as the local network is configured to allow source specific
   joins outside the suggested SSM range. In order for a host to
   receive traffic from an SSM capable source, it must support the
   IGMPv3 multicast group membership reporting protocol, which enables
   the host to explicitly request traffic from a (source,group) pair.
   An SDP syntax is defined in Section 10 to specify the mode of
   operation for the session and the session characteristics such as
   the (source, group) identifier and feedback address.

   The modifications proposed in this document are intended to
   supplement the existing RTCP feedback mechanisms described in [1],
   Section 6. For certain distribution networks, such as SSM networks,
   this may be a requirement, whereas in others it is an optional
   feature that may be used.


3. Basic Operation
   This draft proposes two new methods to enable receiver feedback to
   all members in a session. Each involves the unicasting of RTCP
   packets to a source whose job it is to re-distribute the information
   to the members of the group. The source must always be able to

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   communicate with all group members in order for either mechanism to
   work.

   The first method, the 'Simple Feedback Model', is a basic mechanism
   whereby all Receiver Reports are unicast to the source and
   subsequently forwarded by the source to all receivers on the
   multicast RTCP channel. The advantage of using this method is that
   an existing receiver implementation requires little modification in
   order to use it. Instead of sending Receiver Reports to a multicast
   address, a receiver uses a unicast address and still receives RTCP
   traffic in the usual manner. This method also has the advantage of
   being backwards compatible with RTP/RTCP implementations that do not
   support unicast feedback to the source and operate using the
   standard multicast group communication model, ASM. In a session that
   is using ASM, such a receiver would multicast Receiver Reports to
   the group address and port+1 as stated in [1]. This would still be
   received by all receivers. In a session using SSM, the network
   prevents any data from the receiver being distributed further than
   the first hop router. Additionally, any data heard from this
   receiver by other hosts on the same subnet should be filtered out by
   the host IP stack and therefore will not cause any problems with
   respect to the calculation of Receiver RTCP bandwidth since this
   receiver will not be heard by any other members.

   The second method, the 'Sender Feedback Summary Model' is a
   summarised reporting scheme that provides savings in bandwidth by
   consolidating all the Receiver Reports into one summary packet that
   is then distributed to all the receivers. The advantage of this
   scheme is apparent for large group sessions where the basic
   forwarding mechanism outlined above would create a large amount of
   packet replication in order to forward all the information to all
   the receivers. The basic operation of the scheme is the same as the
   first method, however it requires that all the members in the
   session understand the new summarised packet format outlined in
   Section 7.1. Additionally, the summarised scheme provides a generic
   mechanism for sending distribution information about the data
   reported by the whole group. Potential uses for this are addressed
   in Section 7.4.

   To differentiate between the two reporting mechanisms, a new SDP
   identifier is created and discussed in Section 10. The method of
   reporting must be decided prior to the start of the session, a
   distribution source may not change the method during a session.


4. Definitions
   Distribution Source: In order for unicast feedback to work, there
   must only be one session distribution source for any subset of
   receivers to which RTCP feedback is directed. Heterogeneous
   networks comprised of ASM multiple sender groups, unicast-only
   clients and/or SSM single-sender receiver groups may be
   connected via translators or mixers (see Section 9 for details) to
   create a single source group. In order for unicast feedback to work,

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   only one source must be responsible for distributing the RTP stream
   and for forwarding RTCP information to all receivers. That source is
   called the distribution source.

   RTP and RTCP Channels: The data distributed from the source to the
   receivers is referred to as the RTP and RTCP channels. These
   channels are differentiated via the port numbers as [port] and [port
   + 1] for RTP and RTCP respectively. See [1] for further explanation
   of the port numbering for these channels.

   Unicast RTCP Feedback Target: For a session defined as having a
   distribution source A, on ports n and n+1, the unicast RTCP feedback
   target is the IP address of Source A on port n+1 unless otherwise
   stated in the SDP setup information. See Section 10 for details on
   how the address is specified.

   SSRC: Synchronization source. A 32-bit value that uniquely
   identifies each member in a session. See [1] for further
   information.

   Report blocks: In RTCP [1], it is encouraged to stack multiple
   report blocks in Sender and Receiver Report packets. In this way, a
   variable size packet is created that can include information from
   one source pertaining to multiple sources in the group. The concept
   of report blocks is extended in this draft to encompass Generic
   Summary Report packets in which a source can optionally stack
   multiple reports into one packet in order to provide additional
   feedback on the RTCP traffic received from the group.


5. Packet types
   The RTCP packet types defined in [1] are:

   type       description              Payload number
   SR         sender report            200
   RR         receiver report          201
   SDES       source description       202
   BYE        goodbye                  203
   APP        application-defined      204

   These remain unmodified. Later profile extensions may be added to
   these which are not covered in [1] or this document. In addition to
   the existing types, two new packet types are introduced. Further
   information on each of these is provided in this draft.

   The new packet types are:

   type       description                    Payload number
   RSI        Receiver Summary Information   [see Section 12]
   GSR        General Summary Report         [see Section 12]

   Within the General Summary Report packet, various types of
   distribution data may be reported, each of which requires a

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   distribution type identifier. Current types addressed in this
   document are:

   Distribution Type            Number
   Packet Loss                  1
   Receiver Jitter              2
   Round Trip Time estimation   3
   SSRC distribution            4


6. Simple feedback model
6.1 Packet Formats
   For this mechanism, the packet types used remain the same as for
   standard RTCP feedback in [1]. Receivers still generate Receiver
   Reports with information on the quality of the stream received from
   the source. The distribution source still must create Sender Reports
   that include timestamp information for stream synchronisation and
   round trip time calculation. Both the senders and receivers are
   required to send SDES packets as outlined in [1]. The rules for
   generating BYE and APP packets as outlined in [1] also apply.

6.2 Distribution Source behaviour
   For the simple feedback model, the source provides a simple packet
   reflection mechanism. It is the default behaviour for any
   distribution source and is the minimum requirement for acting as a
   source to a group of receivers using unicast RTCP feedback.

   The source may not stack report blocks received from different SSRCs
   into one packet for retransmission to the group. Every RTCP packet
   from each receiver must be reflected individually.

   The source must listen for unicast RTCP data sent to the RTCP port.
   All unicast data received on this port must be forwarded to the
   group on the multicast RTCP channel. Any multicast data received on
   this port must not be forwarded but processed as defined in [1].

   The reflected traffic should not be included in the transmission
   interval calculation by the source. In other words, the source
   should not consider reflected packets as part of it's own control
   data bandwidth allowance. The algorithm for computing the allowance
   is explained in Section 9. The control bandwidth traffic included in
   the calculation includes any Sender reports to the group, along with
   any additional SDES and APP packets.

   If an application wishes to use APP packets, it is recommended that
   the 'Simple Feedback Model' be used since it is likely that all
   receivers in the session will need to hear the APP specific packets.
   The same applies for all other future RTCP packets that are not
   defined in the base RTP specification [1]. This decision must be
   made in advance of the session and indicated in the SDP
   announcement.

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6.3 Receiver behaviour
   Receivers listen on the RTP and RTCP channels for data. Each
   receiver calculates its share of the receiver bandwidth based on the
   standard rules, i.e., 75% of the RTCP bandwidth is divided equally
   between all unique SSRCs in the session. See Section 9 for further
   information on the calculation of the bandwidth allowance. When a
   receiver is eligible to transmit, it sends a unicast Receiver Report
   packet to the RTCP port of the distribution source.

7. Sender feedback summary model
   In the sender feedback summary mode, the sender is required to
   summarise the information received from all the Receiver Reports
   generated by the receivers and place the information into summary
   reports. The sender feedback summary model introduces two new
   packets. The Receiver Summary Information packet (RSI) which must be
   sent by a source if the summarised feedback mechanism is selected
   and the optional General Summary Report packet (GSR) that may be
   appended to the RSI packet to provide more detailed information on
   the overall session characteristics reported by all receivers.

   The sender must send at least one Receiver Summary Information
   packet for each reporting interval. The sender can additionally
   stack General Summary Reports(GSRs) after the RSI packet. Each GSR
   packet corresponds to the initial RSI packet and acts as an
   enhancement to the basic summary information required by the
   receivers to calculate their reporting time interval. For this
   reason, GSR packets are not required but recommended. RSI and GSR
   packets are sent in addition to the standard Sender Reports and SDES
   packets outlined in [1].

7.1 Packet Formats


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7.1.1 RSI: Receiver Summary Information RTCP Packet
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|    SC   |      PT       |             length            |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                         SSRC of Sender                        |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                           Timestamp                           |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|                           group size                          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|      AFL      |                    HCNL                       |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                   Highest interarrival jitter                 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                    Receiver RTCP Bandwidth                    |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                       collision SSRC #1                       |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                             . . .                             |


   The RSI packet consists of a main report block modeled along the
   same lines as a Receiver Report with optional GSR blocks appended.
   The first eight bytes of header extension follow the standard RTP
   header outline. This ensures backwards compatibility with older
   versions that may not understand the RSI packet format but can read
   the length field indicating the end of the report block. The
   following fields are included:

   The fields "V", "P", and "length" have the same meaning as per [1].

   SC: 5 bits
      The number of collision SSRC entries towards the end of the
      report block. A value of 0 is allowed, indicating that no
      collisions are reported.

   SSRC: 32 bits
      The synchronisation source identifier for the originator of the
      summary report packet.

   timestamp: 32 bits
      The time the packet was sent. This is an unsigned integer value
      displayed in NTP timestamp units to enable detection of duplicate
      packets, reordering and to provide a chronological profile of the
      feedback reports.

   group size: 32 bits
      This field provides the sender's view of the number of receivers
      in a session. This should include the sender itself and any other
      senders potentially connected to the session e.g. via a
      mixer/translator gateway. The group size is calculated according
      to the rules outlined in [1].

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   average fraction lost (AFL): 8 bits
      The average fraction lost indicated by Receiver Reports forwarded
      to this source, expressed as a fixed point number with the binary
      point at the left edge of the field.

   highest cumulative number of packets lost (HCNL): 24 bits
      Highest 'cumulative number of packets lost' value out of all RTCP
      RR packets since the last RSI from any of the receivers.

   highest interarrival jitter: 32 bits
      Highest 'interarrival jitter' value out of all RTCP RR packets
      since the last RSI from any of the receivers.

   receiver bandwidth: 32 bits
      indicates the maximum bandwidth allocated to any single receiver
      for sending RTCP data relating to the session. This is a fraction
      value indicating a percentage of the session bandwidth, expressed
      as a fixed point number with the binary point at the left edge of
      the field.

   collision SSRC: n x 32 bits
      the final fields in the packet are used to identify any SSRCs
      that are duplicated within the group. Usually this is handled by
      the hosts upon detection of the same SSRC, however since
      receivers no longer have a global view of the session, the
      collision algorithm is handled by the source. SSRCs that collide
      are listed in the packet and it is the responsibility of the
      receiver(s) to detect the collision and select another SSRC.

7.1.2 GSR: General Summary Report RTCP Packet Header
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|    BC   |      PT       |            Length             |header
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                         SSRC of Sender                        |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|                             ...                               |report
|                             ...                               |blocks


   The GSR packet is a three-level structure composed of a header and
   zero or more report blocks, each of which describes a range of
   distribution values. The report blocks are a variable length, with a
   fixed header and are described in subsequent sections.

   The fields "V", "P", and "length" have the same meaning as per [1].

   block count (BC): 5 bits
      The number of report blocks contained in this packet

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   SSRC of Sender: 32 bits
      The SSRC of the distribution source


7.1.3 GSR Report block


 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|       DT      |          NDB          |   MF  |     Length    |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                   Minimum Distribution Value                  |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                   Maximum Distribution Value                  |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
|                      Distribution Buckets                     |
|                             ...                               |
|                             ...                               |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+



   distribution type (DT): 8 bits
      A numeric identifier to indicate the significance of the
      distribution values. The items currently defined are described in
      the next sections. Additional items may be defined in a separate
      profile by registering the type numbers with IANA, see Section
      12.

   number of distribution buckets (NDB): 12 bits
      The number of distribution Buckets within the data. The size of
      the bucket can be calculated using the formula, number of bits
      equals (length * 4 * 8)/NDB. Providing 12 bits enables bucket
      sizes as small as 2 bits for a full length packet. The bucket
      size in bits must always be divisable by 2 to ensure byte
      alignment. A bucket size of 2 bits is fairly restrictive,
      however, and it is expected that larger bucket sizes will be more
      practical for most distributions.

   multiplicative factor (MF): 4 bits
      Indicates the multiplicative factor to be applied to each
      distribution Bucket value. Possible values are 1 - 15.

   length: 8 bits
      The length of the whole GSR data packet in 4 byte units. The full
      length of the packet in bytes is calculated by multiplying the
      length value by 4. This tells the receiver the full length of the
      packet and enables the receiver to identify the bucket size. The
      maximum data portion of the packet therefore may be 1008 bytes
      which would provide up to 4032 data buckets of length 2 bits, or
      2016 data buckets of length 4 bits etcà.


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   minimum distribution value: 32 bits
      The Minimum distribution value, in combination with the Maximum
      Distribution value, indicates the range covered by the Data
      Bucket values.

   maximum distribution value: 32 bits
      The Maximum distribution value, in combination with the Minimum
      Distribution value, indicates the range covered by the Data
      Bucket values.

   distribution buckets: each bucket is((length * 4) û 12)*8/NDB bits
      The size and number of buckets depends upon the value of NDB and
      the length of the packet. In order to calculate the size of the
      bucket, the formula ((length * 4) û 12)*8/NDB should be used.
      This indicates the division of the data space and the size of
      each data point in bits. Each value must be multiplied by the
      multiplicative factor.

   Interpretation of the minimum, maximum and distribution values in
   the report block are profile-specific and are addressed
   individually. The size of the report block is variable, as indicated
   by the packet length field.


7.1.4 GSR Loss report block
   GSR loss report blocks indicate the distribution of losses as
   reported by the receivers to the distribution source. Values are
   expressed as a fixed point number with the binary point at the left
   edge of the field. The distribution type is 1.

   Valid results for the Minimum Distribution Value field are 0 - 99,
   otherwise interpreted as 0 - 0.99. Similarly, Valid results for the
   maximum distribution value field are 1 - 100, otherwise interpreted
   as 0.1 - 1. The Minimum Distribution Value must always be less than
   the maximum.

   For examples on processing GSR loss report blocks, see the Appendix.


7.1.5 GSR Jitter report block
   GSR jitter report blocks indicate the distribution of the estimated
   statistical variance of the RTP data packet interarrival time
   reported by the receivers to the distribution source. See [1] for
   details on how the values are calculated and the relevance of the
   jitter results. Jitter values are measured in timestamp units and
   expressed as unsigned integers. The Minimum Distribution Value must
   always be less than the maximum. The distribution type is 2.

7.1.6 GSR Round Trip Time report block
   GSR round trip time reports indicate the distribution of round trip
   times from the distribution source to the receivers. The

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   distribution source is the only member of the group capable of
   calculating the round trip time to any other members since it is the
   only sender in the group. The sender has the option of distributing
   these round trip time estimations to the whole group, uses of which
   are described in Section 7.4. Round trip times are measured in
   timestamp units and expressed as unsigned integers. The
   multiplicative factor can be used to reduce the number of bits
   required to represent the values. The Minimum Distribution Value
   must always be less than the maximum. The distribution type is 3.

7.1.7 SSRC Distribution report block
   SSRC Distributions are an optional feature that can be provided by
   the distribution source to indicate the allocation of SSRCs across
   the group. SSRCs are expressed as unsigned integers. The
   multiplicative factor can be used to reduce the number of bits
   required to represent the values. The Minimum Distribution Value
   must always be less than the maximum. The distribution type is 4.

7.2 Distribution Source behaviour
   The length field of the RSI packet must be calculated over the
   length of the whole RSI packet, using the method defined in [1]. The
   group size must be included in the RSI packet. The source should
   also calculate the Receiver RTCP bandwidth field. Typically this
   value will be calculated as outlined in [1] using the group size and
   session bandwidth as variables. This field however does provide the
   source with the capability to control the amount of feedback from
   the receivers and can be increased or decreased based on the
   requirements of the source. Regardless of the value selected by the
   source for the RTCP bandwidth field, the source must continue to
   forward Sender reports and RSI packets at the rate allowed by its
   bandwidth allocation. See Section 9 for further details.

   In order to identify SSRC collisions, the source is responsible for
   maintaining a record of each SSRC and the corresponding CNAME within
   at least one reporting interval in order to differentiate between
   clients. It is recommended that an updated list of more than one
   interval be maintained to increase accuracy. This mechanism should
   prevent the possibility of collisions since the combination of SSRC
   and CNAME should be unique if the CNAME is generated correctly. In
   the event that collisions are not detected, the effect will be an
   inaccurate impression of the group size on the part of the source.
   Since the statistical probability that collisions will both occur
   and be undetectable is very low, the clients would have to randomly
   select the same SSRC and have the same username + IP address (e.g.
   using private address space behind a NAT router), this should not be
   a significant concern.

   For the GSR packet, the source must decide which are the most
   significant feedback values to convey. The packet format provides
   flexibility in the amount of detail conveyed by the data points.
   There is a trade-off between the granularity of the data and the
   accuracy based on the factorisation values, the number of buckets

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   and the min and max values. In order to focus on a particular region
   of the distribution, the source can adjust the minimum and maximum
   values and either increase the number of buckets and possibly the
   factorisation, or decrease the number of buckets and provide more
   accurate values. See Appendix B for detailed examples on how to
   convey information in RTCP Receiver Reports as GSR information.

   The results should correspond as near as possible to the values
   received during the interval since the last report. The source may
   stack as many report blocks as required in order to convey different
   distributions. If the distribution size exceeds the largest packet
   length (1008 bytes data portion), more packets may be stacked with
   additional information up to the MTU of the connection.


7.3 Receiver behaviour
   The receiver must process RSI packets and adapt session parameters
   such as the RTCP bandwidth based on the information received. The
   receiver no longer has a global view of the session, and will
   therefore be unable to receive information from individual receivers
   aside from itself. However, the information portrayed by the source
   can be extremely detailed, providing the receiver with an accurate
   view of the session quality overall, without the processing overhead
   associated with listening to and analysing all the Receiver Reports.

   The SSRC collision list must be checked against the SSRC selected by
   the receiver to ensure there are no collisions. The group size value
   provides the receiver with the data necessary to calculate it's
   share of the RTCP bandwidth. This share of the bandwidth may be
   overridden by the 'Receiver RTCP Bandwidth' field. This field
   provides the source with the capability to control the amount of
   feedback from the receivers.

   The receiver can handle the GSR data as desired. This data is most
   useful in providing the receiver with a more global view of the
   conditions experienced by other receivers, and enables the client to
   place itself within the distribution and establish the extent to
   which it's reported conditions correspond to the group reports as a
   whole. Appendix A provides further information and examples of data
   processing at the receiver.

   The receiver should assume that any report blocks in the same packet
   correspond to the same data set received by the source during the
   last reporting time interval. This applies to packets with multiple
   blocks, where each block conveys a different range of values.


7.4 Analyzing summarised reports
   Providing a distribution function in a feedback message has a number
   of uses for different types of applications. Although this section
   enumerates potential uses for the distribution scheme, it is

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                        RTCP with Unicast Feedback

   anticipated that future applications might benefit from it in ways
   not addressed in this document. Due to the flexible nature of the
   GSR packet format, future extensions may easily be added. Some of
   the scenarios addressed in this section envisage potential uses
   beyond a simple SSM architecture. For example, single-source group
   topologies where every receiver may in fact also be capable of
   becoming the source. Another example may be multiple SSM topologies
   which combined make up a larger distribution tree.

   A distribution function is useful as input into any algorithm,
   multicast or otherwise, that could be optimized or tuned as a result
   of having access to the feedback values for all group members.

   Following is a list of example areas that might benefit from
   distribution information:

   - The parameterization of a multicast Forward Error Correction (FEC)
   algorithm. Given an accurate estimate of the distribution of
   reported losses, a source or other distribution agent, which does
   not have a global view, would be able to tune the degree of
   redundancy built in to the FEC stream. The distribution might help
   to identify whether the majority of the group is experiencing high
   levels of loss, or whether in fact the high loss reports are only
   from a small subset of the group. Similarly, this data might enable
   a receiver to make a more informed decision about whether it should
   leave a group when it is a very high percentage of the worst case
   reporters.

   - The organization of a multicast data stream into useful layers for
   layered coding schemes. The distribution of packet losses and delay
   would help to identify what percentage of members experience various
   loss and delay levels, and thus how the data stream bandwidth might
   be partitioned to suit the group conditions.

   - The establishment of a suitable feedback threshold. An application
   might be interested to generate feedback values when above (or
   below) a particular threshold.  However, determining an appropriate
   threshold may be difficult when the range and distribution of
   feedback values is not known a priori.  In a very large group,
   knowing the distribution of feedback values would allow a reasonable
   threshold value to be established, and in turn would have the
   potential to prevent message implosion if many group members share
   the same feedback value. A typical application might include a
   sensor network that gauges temperature or some other natural
   phenomenon.  Another example is a network of mobile devices
   interested in tracking signal power to assist with hand-off to a
   different distribution device when power becomes too low.

   - The tuning of Suppression algorithms. Having access to the
   distribution of round trip times, bandwidth, and network loss would
   allow optimization of wake-up timers and proper adjustment of the
   Suppression interval bounds.  In addition, biased wake-up functions
   could be created not only to favor the early response from more

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                        RTCP with Unicast Feedback

   capable group members, but also to smooth out responses from
   subsequent respondents and to avoid bursty response traffic.

   - Leader election among a group of processes based on the maximum or
   minimum of some attribute value. Knowledge of the distribution of
   values would allow a group of processes to select a leader process
   or processes to act on behalf of the group. Leader election can
   promote scalability when group sizes become extremely large.


8. Mixer/Translator issues
   The original RTP specification allows for the use of mixers and
   translators in an RTP session which help to connect heterogeneous
   networks into one session. There are a number of issues, however,
   which are raised by the unicast feedback model proposed in this
   document. The term 'mixer' refers to devices that provide data
   stream multiplexing where multiple sources are combined into one
   stream. Conversely, a translator does not multiplex streams, but
   simply acts as a bridge between two distribution mechanisms, e.g., a
   unicast-to-multicast network translator. Since the issues raised by
   this draft apply equally to either a mixer or translator, they are
   referred to from this point onwards generically as a gateway.

   A gateway between distribution networks in a session must ensure
   that all members in the session receive all the relevant traffic to
   enable the usual operation by the clients. A typical use may be to
   connect an older implementation of an RTP client with an SSM
   distribution network, where the client is not capable of unicasting
   feedback to the source. In this instance the gateway must join the
   session on behalf of the client and send and receive traffic from
   the session to the client. Certain hybrid scenarios may have
   different requirements.


8.1 Use of a mixer-translator
   The gateway must adhere to the SDP descriptor for the single source
   session and use the feedback mechanism indicated. Receivers should
   be aware that by introducing a gateway into the session, more than
   one source may potentially be active in a session since the gateway
   may be forwarding traffic from either multiple unicast sources or
   from an ASM session to the SSM receivers. Receivers should still
   forward unicast RTCP reports in the usual manner to the distribution
   source, which in this case would be the gateway itself. It is
   recommended that the simple packet reflection mechanism be used
   under these circumstances since attempting to coordinate RSI + LJS
   reporting between more than one source may be complicated unless the
   gateway is capable of undertaking the summarisation itself.



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                        RTCP with Unicast Feedback

8.2 Encryption and Authentication issues
   Encryption and security issues are discussed in detail in Section
   11. A gateway must be able to follow the same security policy as the
   client in order to unicast forward RTCP data to the source, and it
   therefore must be able to apply the same authentication and/or
   encryption policy required for the session. Transparent bridging,
   where the gateway is not acting as the distribution source, and
   subsequent unicast feedback to the source is only allowed if the
   gateway can conduct the same source authentication as required by
   the receivers.


9. Transmission interval calculation
   The Control Traffic Bandwidth referred to in [1] is an arbitrary
   amount which is intended to be supplied by a session management
   application (e.g., [9]) or decided based upon the bandwidth of a
   single sender in a session. A receiver must calculate the number of
   other members in a session based upon either its own SSRC count
   determined by the forwarded Receiver Reports, or from the RSI report
   from a sender.

   The RTCP transmission Interval calculation remains the same as in
   the original RTP specification [1]. In the original specification,
   the senders are allocated 1/4 of the control traffic bandwidth if
   they number 25% or less than the group size. Otherwise the
   allocation for senders is the percentage of senders to group size.
   The remaining bandwidth is allocated to the receivers to be divided
   evenly amongst the group. The source should calculate the
   transmission interval for RSI + LJS packets out of its 1/4 of the
   control traffic bandwidth with a minimum transmission interval of 5
   seconds.


10. SDP Extensions
   The Session Description Protocol (SDP) is used as a means to
   describe media sessions in terms of their transport addresses,
   codecs, and other attributes. Providing RTCP feedback via unicast as
   specified in this document constitutes another session parameter
   needed in the session description. Similarly, parameters of SSM RTCP
   feedback -- such as the mode of summarizing information at the
   sender and the target unicast address to which to send feedback
   information -- need to be provided.  This section defines the SDP
   parameters that are needed by the proposed mechanisms in this draft
   (and that also need to be registered with IANA).


10.1 SSM RTCP Session Identification
   A new session level attributes MUST be used to indicate the use of
   unicast instead of multicast feedback: "rtcp:unicast".


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                        RTCP with Unicast Feedback

   This attribute uses one additional parameter to specify the mode of
   operation.

   rtcp:unicast reflection -- MUST be used to indicate packet
   reflection by the RTCP target (without further processing).

   rtcp:unicast gsr        -- MUST be used to indicate the "General
   Summary Report" mode of operation.

   rtcp:unicast rsi        -- MUST be used to indicate the "Receiver
   Summary Information" mode of operation.


10.2 SSM Source Specification
   In addition, in an SSM RTCP session, the sender(s) need to be
   indicated for both source-specific joins to the multicast group as
   well as for addressing RTCP packets to.

   This is done following the proposal for SDP source filters
   documented in draft-ietf-mmusic-sdp-srcfilter-00.txt [15].

   From this specification, only the inclusion mode ("a=incl:") MUST be
   used for SSM RTCP.

   There SHOULD be exactly one "a=incl:" attribute listing the address
   of the sender.  The RTCP port MUST be derived from the m= line of
   the media description.

   An optional alternative feedback address may be supplied using an
   attribute such as a=rtcp:<port> IN IP4 192.168.1.1.


11. Security Considerations

11.1 Assumptions

   RTP/RTCP is a protocol for carrying real-time multimedia traffic,
   and therefore one of the main considerations for any security
   solution must be to maintain as low an overhead as possible in order
   to limit processing constraints. This includes the consideration of
   overhead for different types of cryptographic operations on data, as
   well as considerations for deploying or creating security
   infrastructure for large groups.

   The distribution of session parameters, typically using SDP type
   information through SAP, email or the web is beyond the scope of
   this document. It is recommended, however, that the method used
   should employ adequate security measures to ensure the integrity and
   authenticity of the information. For the purposes of this analysis,
   it is assumed that the information has already been securely
   distributed out-of-band.

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                        RTCP with Unicast Feedback


   It is also assumed that the multicast or group distribution
   mechanism e.g. the SSM routing tree, is not immune to source IP
   address spoofing or traffic snooping. All security weaknesses are
   therefore addressed from a transport level perspective or above.

11.2 Security threats

   Attacks on this architecture may take a variety of forms, and in
   order to identify the security weaknesses, it is important to
   address these individually.

   a) Denial of Service
      A major area of concern would be a distributed denial of service
      attack. Due to the nature of the communication architecture this
      is a situation that could be generated a number of ways by using
      the unicast feedback characteristic as a weakness. Since standard
      multicast communication does not typically involve many to one
      unicast forwarding of data, this poses new challenges for a
      security solution.

   b) Packet Forgery
      One potential area of attack to guard against is packet forgery.
      In particular, it is important to protect against the integrity
      of certain influential packets since compromise of certain
      control packets could directly affect the transmission
      characteristics of the whole group, however for the purposes of
      defining a security profile, every packet is considered equally
      as important. In the case of a large group, the compromise of
      RTCP traffic could have serious consequences.

   c) Session Replay
      An additional concern is the potential for session recording and
      subsequent replay. The issue to deal with in particular in this
      instance is that an attacker may not actually need to understand
      the packet contents, but just simply have the ability to record
      the data stream and at a later time replay it with a spoofed
      source address.

   d) Eavesdropping on a session
      The consequences of eavesdropping by an attacker on a session may
      not directly constitute a security weakness, however it might
      benefit other types of attack, and should therefore be considered
      as a potential threat to guard against.


11.3 Security properties

   Three types of security that may be applied to combat the issues
   identified above seem relevant for these contexts.

   a) Data integrity

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                        RTCP with Unicast Feedback

      This ensures that the data received from the network has not been
      tampered with by any third party, either maliciously or through a
      network error. This type of test ensures that the packet received
      is guaranteed to be in exactly the same condition as that source
      intended it to be. This does not guard against the authenticity
      of the source that created the packet.

   b) Data authenticity
      In order to determine what entity is generating certain data, an
      authenticity mechanism is required. This guarantees that the
      creator of the data is known to the receiver and that the
      receiver can trust the content of the data assuming the data
      integrity has also been secured.

   c) Data confidentiality
      In order to restrict information access to authorized entities,
      confidentiality may also be required. This ensures that only
      authorized clients can understand the data that they receive. It
      does not prevent eavesdroppers receiving the traffic and having
      the capability to replay information in it's original form to
      other clients with the capability to understand the information.

   d) Replay protection
      Ensures that given some pre-determined range of either time or
      session values, a host can determine whether the data was
      transmitted within the given window.


11.4 Architectural Contexts

   In order to understand the potential weaknesses to guard against, it
   is necessary to divide the communication model into a number of
   distinct contexts.

   a) Source to Receiver communication
      The first, and perhaps most influential context to protect, is
      the ædownstreamÆ communication channel from the source to the
      receivers. This is effectively the main controlling influence
      over the behaviour of the group since it determines the bandwidth
      allocation for each receiver and hence the rate at which the RTCP
      traffic is directly unicast back to the source. All traffic that
      is distributed over the downstream channel should be generated by
      a single source. Both the RTP data stream and the RTCP control
      data are sent over this channel. The RTCP data is indirectly
      influenced by the information the source has received from the
      whole group.

      This context is vulnerable to all four attacks outlined in the
      previous section. A denial of service attack from the source to
      the receivers is possible, but less of a concern since the worst
      case effect of sending large volumes of traffic over the
      distribution channel has the potential to reach every receiver,

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                        RTCP with Unicast Feedback

      but only on a one-to-one basis, this is no different from the
      current multicast model where an individual source may send large
      volumes of traffic to a multicast group. The real danger of
      denial of service attacks in this context comes indirectly via
      compromise of the source RTCP traffic. If receivers are provided
      with an incorrect group size estimate or bandwidth allowance, the
      return traffic to the source may create a Distributed DoS effect
      on the source. Similarly, an incorrect feedback address whether
      as a result of a malicious attack or by mistake e.g. an IP
      address typing error, could directly create a denial of service
      attack on another host which must also be guarded against.

      The danger of Packet forgery in the worst case may be to
      maliciously instigate a denial of service attack, e.g. if an
      attacker were capable of spoofing the source address and
      injecting incorrect packets into the data stream or intercepting
      the source RTCP traffic and modifying the fields. Other
      consequences of packet forgery in this context may be the
      compromise of data affecting the integrity of the data received
      both in the RTP stream itself and the RTCP data in general.

      The replay of a session would have the effect of recreating the
      receiver feedback to the source address at a time when the source
      is not expecting additional traffic from a potentially very large
      group. The consequences of this type of attack may be less
      effective on their own, but in combination with other attacks
      might be serious.

      Eavesdropping on the session would provide an attacker with
      information on the charateristics of the source to receiver
      traffic such as the frequency of RTCP traffic and, if
      unencrypted, might also provide valuable information on
      characteristics such as group size and transmission
      charateristics of the receivers back to the source in addition to
      enabling an attacker to listen to the media streams. In this
      context, the attacker might also have access to personal
      information carried in the SDES packets such as email, phone and
      full username information.

   b) Receiver to source or gateway communication
      The second context to address is the return traffic from the
      group to the source or gateway which for the purposes of this
      analysis may be considered in the same light as a distribution
      source. This traffic should only be RTCP type data, and should
      include receiver reports, SDES information and possibly
      Application specific packets. The effects of compromise on a
      single or subset of receivers is less likely to have as great an
      impact as the first context, however much of the responsibility
      for detecting compromise of the source data stream relies on the
      receivers.

      The effects of compromise of the first context with respect to
      critical source RTCP control information would be witnessed most
      seriously in the second context. A large group of receivers may

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                        RTCP with Unicast Feedback

      unwittingly generate a distributed DoS attack on the source in
      the event that the intergrity of the source RTCP channel has been
      compromised and the security breach is not detectable by the
      individual receivers.

      In the event that packet forgery may occur in this context, the
      effect may be the introduction of false RTCP traffic and/or the
      creation of fake SSRC identifiers. Such an attack might slow down
      the overall control channel data rate, since an incorrect
      perception of the group size may be created. This might affect
      external issues such as group accounting and other as yet unknown
      potential uses of the distribution functionality for controlling
      group behaviour such as leader election based on feedback
      criteria.

      A replay attack on receiver return data to the source would have
      the same implications as the generation of false SSRC identifiers
      and RTCP traffic to the source. It is therefore equally as
      important to protect against compromise of any receiver
      contribution to the RTCP channel as it is to ensure authenticity
      and freshness of the data source.

      Eavesdropping in this context may potentially provide an attacker
      with a great deal of personal information about a large group of
      receivers available from SDES packets. It would also provide an
      attacker with information on group traffic generation
      characteristics and parameters for calculating individual
      receiver bandwidth allowance.


11.5 Requirements in each context

   Some initial requirements to consider for each context in general
   are that the overhead of ensuring the security of the session should
   be kept as low as possible. This entails keeping the setup and
   communication of shared or private keys to a minimum. The nature of
   RTP/RTCP traffic is that sessions require real-time processing and
   minimal overhead for communication. This means that processing
   constraints imposed by techniques such as public/private key
   encryption versus stream ciphers using shared keys are an important
   consideration for defining this security profile.

   Having identified the security weaknesses for each communication
   context, security type requirements can be addressed for each.

   a) The first context is concerned with denial of service attacks
      through possible packet forgery. The forgery may take the form of
      interception and modification of packets from the source, or
      simply injecting false packets into the distribution channel. To
      combat these attacks, data integrity and source authenticity are
      required. The degree of confidentiality which may be deployed is
      not a requirement in this context since the actual consequences
      of eavesdropping do not affect the operation of the protocol,

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                        RTCP with Unicast Feedback

      however without confidentiality, access to personal and group
      characteristics information would be unrestricted to an external
      listener and it is therefore recommended.

   b) The second context must defend against the same kinds of attacks.
      Data integrity is required to ensure that interception and
      modification of an individual receiverÆs RTCP traffic is not
      accomplished. This is to protect against the false influence of
      group control information and the possible serious compromise of
      future services provided by the distribution functionality such
      as leader election based on various parameters. In order to
      ensure data integrity, receiver authenticity is therefore an
      additional requirement in order to ensure the origin of the data
      is secure. The same situation applies as in the first context
      with respect to data confidentiality, and it is recommended that
      precautions should be taken to protect the privacy of the data.


11.6 Overview of existing security solutions

   This section addresses some existing group security mechanisms and
   identifies which aspects of the security requirements they might
   provide. This security analysis is a work in progress, and these
   options will be explored in more detail in subsequent versions of
   the draft.

   SRTP provides confidentiality of the RTP and RTCP packets as well as
   protection against integrity compromise and replay attacks. It
   provides authentication of the data stream, however it does not
   provide authentication on a per-user basis. This means that a packet
   can be authenticated as having originated from one of the session
   members, but it does not indicate which member. All keys for an SRTP
   session are derived from a single master key which it is assumed has
   been distributed via some out-of-band secure method.

   A more general group security profiles which should be considered
   are the Group Domain of Interpretation which provides a solution for
   multicast IPSec ESP security with group authentication.

   GSAKMP is perhaps the most detailed solution which provides group
   access control, key generation and facilities for rekeying the whole
   group.

   These options will be considered individually in later releases of
   the draft.

12. IANA Considerations
   Based on the guidelines suggested in [10], this document proposes 2
   new RTCP data payload types for consideration by IANA, and 4 new

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                        RTCP with Unicast Feedback

   sub-payload types for summary distribution types, defined in section
   5.

   Furthermore, four new SDP media-level attributes are defined in
   Section 10.

13. Outstanding Issues

   6.2 Complication with detecting unicast versus multicast transmitted
   data on the same port.

   Add Backwards compatibility section

   Include implications of recent changes to port/port+1 rules for
   RTP/RTCP traffic.


14. References
   [1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP -
   A Transport Protocol for Real-time Applications," Internet
   Draft, draft-ietf-avt-rtp-new-10.txt, Work in Progress, July 2001.

   [2] Pusateri, T, "Distance Vector Multicast Routing Protocol",
   draft-ietf-idmr-dvmrp-v3-10, August 2000

   [3] Fenner, B, Handley, M, Holbrook, H, Kouvelas, I, "Protocol
   Independent Multicast - Sparse Mode (PIM-SM): Protocol Specification
   (Revised)", draft-ietf-pim-sm-v2-new-02.txt, March 2001

   [4] Farinacci, D, Kouvelas, I, Windisch, K, "State Refresh in PIM-
   DM" draft-ietf-pim-refresh-02.txt, November, 2000

   [5] Thaler, D, Cain, B, "BGP Attributes for Multicast Tree
   Construction", draft-ietf-idmr-bgp-mcast-attr-00.txt, February 1999

   [6] Farinacci, D, Rekhter, Y, Meyer, D, Lothberg, P, Kilmer, H,
   Hall, J, "Multicast Source Discovery Protocol (MSDP)", draft-ietf-
   msdp-spec-06.txt, July 2000

   [7] Shepherd, G, Luczycki, E, Rockell, R, "Source-Specific Protocol
   Independent Multicast in 232/8", draft-shepherd-ssm232-00.txt, March
   2000.

   [8] Holbrook, H, Cain, B, "Using IGMPv3 For Source-Specific
   Multicast", draft-holbrook-idmr-igmpv3-ssm-00.txt, July 2000.

   [9] Session Directory Rendez-vous (SDR), developed at University
   College London by Mark Handley and the Multimedia Research Group.

   [10] Alvestrand, H. and T. Narten, "Guidelines for Writing an IANA
   Considerations Section in RFCs", BCP 26, RFC 2434, October 1998.


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                        RTCP with Unicast Feedback

   [11] Handley, M, Perkins, C, Whelan, E, "Session Announcement
   Protocol", (SAP), RFC 2974, October 2000.

   [12] A. Frier, P. Karlton, and P. Kocher, "The SSL 3.0 Protocol",
   Netscape Communications Corp., Nov 18, 1996.

   [13] Perrig, Canetti, Briscoe, Tygar, Song, "TESLA: Multicast Source
   Authentication Transform", draft-irtf-smug-tesla-00.txt.

   [14] E. Carrara, D. McGrew, M. Naslund, K. Norrman, D. Oran, "The
   Secure Real Time Transport Protocol", draft-ietf-avt-srtp-01.txt.

   [15] B. Quinn, "SDP Source-Filters", Internet Draft draft-ietf-
   mmusic-sdp-srcfilter-00.txt, Work in Progress, May 2000.


15. Appendix
A  GSR packet processing at the receiver

A.1 Algorithm
   Example processing of Loss Distribution Values
   X values represent the loss percentage.
   Y values represent the number of receivers.

   Number of x values is the NDB value
   xrange = Max Distribution Value(MaDV) - Min Distribution Value(MnDV)
   First data point = MnDV,first ydata
   then
   Foreach ydata => xdata += (MnDV + (xrange / NDB))
A.2 Pseudo-code
   Packet Variables -> factor,NDB,MnDVL,MaDV
   Code variables -> xrange, ydata[NDB],x,y

   xrange = MaDV - MnDV
   x = MnDV;



B GSR packet creation at the source
   See Postscript version.

C AUTHORS ADDRESSES
   Julian Chesterfield
   AT&T Labs - Research
   75 Willow Road
   Menlo Park, CA 94025
   julian@research.att.com

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                        RTCP with Unicast Feedback


   Eve Schooler
   AT&T Labs - Research
   75 Willow Road
   Menlo Park, CA 94025
   schooler@research.att.com

   Joerg Ott
   Tellique Kommunikationstechnik GmbH
   Berliner Str. 26
   D-13507 Berlin
   GERMANY
   Phone: +49.30.43095-560  (sip:jo@tzi.org)
   Fax:   +49.30.43095-579
   Email: jo@tellique.com

D FULL COPYRIGHT STATEMENT
   Copyright (C) The Internet Society (2000). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph
   are included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
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   The limited permissions granted above are perpetual and will not be
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   This document and the information contained herein is provided on an
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   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
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   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
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